Digital modulation schemes are increasingly employed to provide high data throughput over inherently band-limited channels. A prominent example of this is the DSL (digital subscriber line) service provided over telephone lines. Another prominent example is wireless internet communication, such as under the Institute for Electrical and Electronics Engineers (IEEE) 802.11 standards.
Since telephone lines are known to provide a narrow-band channel, digital modulation schemes are often employed to increase the data rate over channels of this type. Quadrature amplitude modulation (QAM) systems, in which orthogonal symbols are transmitted, have been very successful in providing high data rates in moderate symbol rate transmissions. In order to optimize the transmission through quadrature schemes over narrow band channels, various types of orthogonal frequency-division multiplexing (OFDM), have been used. In an OFDM system, the available bandwidth is divided into subcarriers that enable easier transmission and reception. The physical layer of IEEE 802.11(a) utilizes such OFDM.
A popular type of OFDM is known as digital multi-tone (DMT). In a DMT system, the total channel bandwidth is divided into subcarriers (frequency bands), but the channel capacity of each subcarrier is generally determined individually during modem or other transceiver training. In other words, each subcarrier may have a different QAM constellation, which allows some subbands to have higher effective data rates than others. As mentioned above, DMT is very similar to the OFDM used in wireless applications, such as in transceivers for IEEE 802.11, and any reference herein to DMT should be understood to mean and include OFDM or any other form of frequency division multiplexing.
One of the advantages of DMT or OFDM is that comparatively less complicated equalization can be employed. Data rates are lower within the subbands, which tends to reduce the overall problem of intersymbol interference (ISI) and thus make the necessary equalizers less complex.
A problem arises in current implementations, however, wherein a number of time-domain equalization techniques are used to derive the coefficients of a short impulse response filter (“SIRF”) that is used to provide the time domain equalization (“TEQ”). One disadvantage is that available channel capacity in such a system is not a smooth function of synchronization delay. In fact, for some particular synchronization delays, channel capacity can be highly degraded. This occurs principally because the TEQ algorithms of present systems do not use individual frequency information, and thus end up treating all tones in the same way.
Consequently, a need arises for an effective equalization system and method that overcomes these problems in the prior art, while maintaining high performance under reasonable complexity, and that may be implemented with existing hardware technologies.